404 not found when trying to make outbound calls

Hi.
Just got skype for business, incoming calls working fine but cannot get outbound to work.
Any ideas?
     Call-ID: YzUwYWZiNDRlMTc0NzhlNDlmYjExMmRjNWFmZDE3M2Q.
     CSeq: 1 INVITE
     Session-Expires: 1800
     Accept: application/sdp, message/sipfrag
     Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY, INFO, REFER, PRACK
     Content-Type: application/sdp
     Supported: replaces, histinfo, 100rel, timer
     User-Agent: TeleVoice Trio Enterprise 3.2 TE 3.2.79
     History-Info: <s
     v=0
     o=- ****************330 1 IN IP4 192.168.0.191
     s=-
     c=IN IP4
     t=0 0
     m=audio 53018 RTP/AVP 8 101 0
     a=rtpmap:8 pcma/8000
     a=rtpmap:101 telephone-event/8000
     a=rtpmap:0 pcmu/8000
     sigcomp id=
 Resiprocate: (level=7)
     Adding message to tx buffer to: [ V4  UDP target domain= mFlowKey=0 ]
 Resiprocate: (level=7)
     incoming from: [ V4  UDP target domain=unspecified mFlowKey=352 ]
 Resiprocate: (level=
     CSeq: 1 INVITE
     Content-Length: 0
 Resiprocate: (level=7)
     incoming from: [ V4 UDP target domain=unspecified mFlowKey=352 ]
 Resiprocate: (level=
     CSeq: 1 INVITE
     Content-Length: 0
 [edited for privacy]

can anyone over at Skype help, there seems to be no online chat support anymore, and I am a paying customer, very dissapointing!
I have Skype Connect  (skype sip) setup for incoming calls and this has been working well for some time, but now i want to call out to skype users using the same account and I get both 407 and 404 errors, a redacted trace copied below:
|Time     | xx.xxx.xx.xxx                         |
|         |                   | 63.209.144.201    |                   
|0.000    |         INVITE SDP (g711U g711A telephone-eventRTPType...1 CN)          |SIP From: "" <sip:[email protected] To:<sip:[email protected]
|         |(5080)   ------------------>  (5060)   |
|0.041    |         100 Trying|                   |SIP Status
|         |(5080)   <------------------  (5060)   |
|0.184    |         407 Proxy Authentication Required          |SIP Status
|         |(5080)   <------------------  (5060)   |
|0.184    |         ACK       |                   |SIP Request
|         |(5080)   ------------------>  (5060)   |
|0.185    |         INVITE SDP (g711U g711A telephone-eventRTPType...1 CN)          |SIP From: "" <sip:[email protected] To:<sip:[email protected]
|         |(5080)   ------------------>  (5060)   |
|0.225    |         100 Trying|                   |SIP Status
|         |(5080)   <------------------  (5060)   |
|0.857    |         404 Not Found                 |SIP Status
|         |(5080)   <------------------  (5060)   |
|0.857    |         ACK       |                   |SIP Request
|         |(5080)   ------------------>  (5060)   |
the (redacted) gateway config in freeswitch is:
<include>
        <gateway name="skype">
        <param name="username" value="9xxxxxxxxxxxxx"/>
        <param name="realm" value="sip.skype.com"/>
         <param name="from-user" value="9xxxxxxxxxxxxx"/>
        <param name="from-domain" value="sip.skype.com"/>
        <param name="password" value="XXXXXXXXXXXXXX"/>
         <param name="extension" value="9xxxxxxxxxxxxx"/>
        <param name="proxy" value="sip.skype.com"/>
        <!-- <param name="register-proxy" value="stun.skype.com"/> -->
        <param name="register" value="true"/>
  <!-- which transport to use for register -->
        <param name="register-transport" value="udp"/>
  <!--How many seconds before a retry when a failure or timeout occurs -->
        <param name="retry-seconds" value="30"/>
        <param name="caller-id-in-from" value="false"/>
        <param name="contact-params" value="[email protected]"/>
        <param name="context" value="public"/>
      </gateway>
</include>
Can someone over at skype support assist? or anyone else who has experience with this?
 The 404 appears to be the more serious error but not sure if the 407 is causing the 404 and needs to be cleared up first.
Thanks!

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